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author | Max Horn | 2003-07-14 20:09:14 +0000 |
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committer | Max Horn | 2003-07-14 20:09:14 +0000 |
commit | 33e6989944f23e2fd7d556a0b6b99c78ac5ac126 (patch) | |
tree | 6bc23b537b42323c23357187817fe666091f1bf5 | |
parent | 66823befe2bafb7bfbab9ec03c7981a3fda2c601 (diff) | |
download | scummvm-rg350-33e6989944f23e2fd7d556a0b6b99c78ac5ac126.tar.gz scummvm-rg350-33e6989944f23e2fd7d556a0b6b99c78ac5ac126.tar.bz2 scummvm-rg350-33e6989944f23e2fd7d556a0b6b99c78ac5ac126.zip |
removed whitespaces at line ends; added stopHandle
svn-id: r9022
-rw-r--r-- | sound/mixer.cpp | 91 | ||||
-rw-r--r-- | sound/mixer.h | 11 |
2 files changed, 63 insertions, 39 deletions
diff --git a/sound/mixer.cpp b/sound/mixer.cpp index c0b81afdae..3eb1592c11 100644 --- a/sound/mixer.cpp +++ b/sound/mixer.cpp @@ -241,7 +241,7 @@ int SoundMixer::insertChannel(PlayingSoundHandle *handle, Channel *chan) { } int SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, int id) { - StackLock lock(_mutex); + StackLock lock(_mutex); // Prevent duplicate sounds if (id != -1) { @@ -254,31 +254,31 @@ int SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, ui } int SoundMixer::newStream(void *sound, uint32 size, uint rate, byte flags, uint32 buffer_size) { - StackLock lock(_mutex); + StackLock lock(_mutex); return insertChannel(NULL, new ChannelStream(this, 0, sound, size, rate, flags, buffer_size)); } #ifdef USE_MAD int SoundMixer::playMP3(PlayingSoundHandle *handle, void *sound, uint32 size, byte flags) { - StackLock lock(_mutex); + StackLock lock(_mutex); return insertChannel(handle, new ChannelMP3(this, handle, _syst->property(OSystem::PROP_GET_SAMPLE_RATE, 0), sound, size, flags)); } int SoundMixer::playMP3CDTrack(PlayingSoundHandle *handle, File *file, mad_timer_t duration) { - StackLock lock(_mutex); + StackLock lock(_mutex); return insertChannel(handle, new ChannelMP3CDMusic(this, handle, _syst->property(OSystem::PROP_GET_SAMPLE_RATE, 0), file, duration)); } #endif #ifdef USE_VORBIS int SoundMixer::playVorbis(PlayingSoundHandle *handle, OggVorbis_File *ov_file, int duration, bool is_cd_track) { - StackLock lock(_mutex); + StackLock lock(_mutex); return insertChannel(handle, new ChannelVorbis(this, handle, ov_file, duration, is_cd_track)); } #endif void SoundMixer::mix(int16 *buf, uint len) { StackLock lock(_mutex); - + if (_premixProc && !_paused) { int i; _premixProc(_premixParam, buf, len); @@ -293,7 +293,7 @@ void SoundMixer::mix(int16 *buf, uint len) { if (!_paused) { // now mix all channels for (int i = 0; i != NUM_CHANNELS; i++) - if (_channels[i]) + if (_channels[i]) _channels[i]->mix(buf, len); } } @@ -315,7 +315,8 @@ bool SoundMixer::bindToSystem(OSystem *syst) { } void SoundMixer::stopAll() { - StackLock lock(_mutex); + StackLock lock(_mutex); +printf("->SoundMixer::stopAll()\n"); for (int i = 0; i != NUM_CHANNELS; i++) if (_channels[i]) _channels[i]->destroy(); @@ -327,13 +328,14 @@ void SoundMixer::stop(int index) { return; } - StackLock lock(_mutex); + StackLock lock(_mutex); +printf("->SoundMixer::stop(%d)\n", index); if (_channels[index]) _channels[index]->destroy(); } void SoundMixer::stopID(int id) { - StackLock lock(_mutex); + StackLock lock(_mutex); for (int i = 0; i != NUM_CHANNELS; i++) { if (_channels[i] != NULL && _channels[i]->_id == id) { _channels[i]->destroy(); @@ -342,6 +344,25 @@ void SoundMixer::stopID(int id) { } } +void SoundMixer::stopHandle(PlayingSoundHandle handle) { + StackLock lock(_mutex); + + // Simply ignore stop requests for handles of sounds that already terminated + if (handle == 0) + return; + + int index = handle - 1; + if ((index < 0) || (index >= NUM_CHANNELS)) { + warning("soundMixer::stopHandle has invalid index %d", index); + return; + } + +printf("->SoundMixer::stopHandle(%d)\n", index); + if (_channels[index]) + _channels[index]->destroy(); +} + + void SoundMixer::pause(bool paused) { _paused = paused; } @@ -351,7 +372,7 @@ bool SoundMixer::hasActiveSFXChannel() { // (and maybe also voice) here to work properly in iMuseDigital // games. In the past that was achieve using the _beginSlots hack. // Since we don't have that anymore, it's not that simple anymore. - StackLock lock(_mutex); + StackLock lock(_mutex); for (int i = 0; i != NUM_CHANNELS; i++) if (_channels[i] && !_channels[i]->isMusicChannel()) return true; @@ -359,14 +380,14 @@ bool SoundMixer::hasActiveSFXChannel() { } bool SoundMixer::isActiveChannel(int index) { - StackLock lock(_mutex); + StackLock lock(_mutex); if (_channels[index]) return _channels[index]->isActive(); return false; } void SoundMixer::setupPremix(void *param, PremixProc *proc) { - StackLock lock(_mutex); + StackLock lock(_mutex); _premixParam = param; _premixProc = proc; } @@ -407,13 +428,13 @@ class CubicInterpolator { protected: int x0, x1, x2, x3; int a, b, c, d; - + public: CubicInterpolator(int a0, int b0, int c0) : x0(2 * a0 - b0), x1(a0), x2(b0), x3(c0) { // We use a simple linear interpolation for x0 updateCoefficients(); } - + inline void feedData() { x0 = x1; x1 = x2; @@ -429,7 +450,7 @@ public: x3 = xNew; updateCoefficients(); } - + /* t must be a 16.16 fixed point number between 0 and 1 */ inline int interpolate(uint32 fpPos) { int result = 0; @@ -438,10 +459,10 @@ public: result = (result * t + c) >> 8; result = (result * t + d) >> 8; result = (result / 3 + 1) >> 1; - + return result; } - + protected: inline void updateCoefficients() { a = ((-x0 * 2) + (x1 * 5) - (x2 * 4) + x3); @@ -470,19 +491,19 @@ static void mix_signed_mono_8(int16 *data, uint &len, byte *&s, uint32 &fp_pos, do { do { result = interp.interpolate(fp_pos); - + *data = clamped_add_16(*data, result); data++; *data = clamped_add_16(*data, result); data++; - + fp_pos += fp_speed; inc = fp_pos >> 16; s += inc; len--; fp_pos &= 0x0000FFFF; } while (!inc && len && (s < s_end)); - + if (s + 2 < s_end) interp.feedData(vol_tab[*(s + 2)]); else @@ -499,12 +520,12 @@ static void mix_unsigned_mono_8(int16 *data, uint &len, byte *&s, uint32 &fp_pos do { do { result = interp.interpolate(fp_pos); - + *data = clamped_add_16(*data, result); data++; *data = clamped_add_16(*data, result); data++; - + fp_pos += fp_speed; inc = fp_pos >> 16; s += inc; @@ -614,15 +635,15 @@ typedef void MixProc(int16 *data, uint &len, byte *&s, uint32 &fp_pos, int fp_speed, const int16 *vol_tab, byte *s_end, bool reverse_stereo); -static MixProc *mixer_helper_table[8] = { - mix_signed_mono_8, mix_unsigned_mono_8, +static MixProc *mixer_helper_table[8] = { + mix_signed_mono_8, mix_unsigned_mono_8, mix_signed_stereo_8, mix_unsigned_stereo_8, - mix_signed_mono_16, mix_unsigned_mono_16, + mix_signed_mono_16, mix_unsigned_mono_16, mix_signed_stereo_16, mix_unsigned_stereo_16 }; static int16 mixer_element_size[] = { - 1, 1, + 1, 1, 2, 2, 2, 2, 4, 4 @@ -727,7 +748,7 @@ void ChannelStream::append(void *data, uint32 len) { if (_endOfData + len > _endOfBuffer) { /* Wrap-around case */ uint32 size_to_end_of_buffer = _endOfBuffer - _endOfData; - uint32 new_size = len - size_to_end_of_buffer; + uint32 new_size = len - size_to_end_of_buffer; if ((_endOfData < _pos) || (_ptr + new_size >= _pos)) { debug(2, "Mixer full... Trying to not break too much (A)"); return; @@ -763,7 +784,7 @@ void ChannelStream::mix(int16 *data, uint len) { return; } - + const int16 *vol_tab = _mixer->_volumeTable; MixProc *mixProc = mixer_helper_table[_flags & 0x07]; @@ -780,7 +801,7 @@ void ChannelStream::mix(int16 *data, uint len) { assert(wrapOffset <= WARP_WORKAROUND); memcpy(_endOfBuffer, _ptr, wrapOffset); } - + mixProc(data, len, _pos, _fpPos, _fpSpeed, vol_tab, _endOfBuffer + wrapOffset, (_flags & SoundMixer::FLAG_REVERSE_STEREO) ? true : false); // recover from wrap @@ -819,7 +840,7 @@ ChannelMP3Common::~ChannelMP3Common() { mad_stream_finish(&_stream); } -ChannelMP3::ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, uint32 rate, void *sound, uint size, byte flags) +ChannelMP3::ChannelMP3(SoundMixer *mixer, PlayingSoundHandle *handle, uint32 rate, void *sound, uint size, byte flags) : ChannelMP3Common(mixer, handle, rate) { _posInFrame = 0xFFFFFFFF; _position = 0; @@ -867,7 +888,7 @@ void ChannelMP3::mix(int16 *data, uint len) { /* Skip _silence_cut a the start */ if ((_posInFrame < _synth.pcm.length) && (_silenceCut > 0)) { uint32 diff = _synth.pcm.length - _posInFrame; - + if (diff > _silenceCut) diff = _silenceCut; _silenceCut -= diff; @@ -969,7 +990,7 @@ void ChannelMP3CDMusic::mix(int16 *data, uint len) { } while (1) { - // Get samples, play samples ... + // Get samples, play samples ... ch = _synth.pcm.samples[0] + _posInFrame; while ((_posInFrame < _synth.pcm.length) && (len > 0)) { int16 sample = (int16)((scale_sample(*ch++) * volume) / 32); @@ -985,7 +1006,7 @@ void ChannelMP3CDMusic::mix(int16 *data, uint len) { } // See if we have finished // May be incorrect to check the size at the end of a frame but I suppose - // they are short enough :) + // they are short enough :) frame_duration = _frame.header.duration; mad_timer_negate(&frame_duration); mad_timer_add(&_duration, frame_duration); @@ -1062,7 +1083,7 @@ void ChannelVorbis::mix(int16 *data, uint len) { #else 0, #endif - 2, 1, + 2, 1, #endif NULL); if (result == 0) { diff --git a/sound/mixer.h b/sound/mixer.h index e5e9149132..abc511ff71 100644 --- a/sound/mixer.h +++ b/sound/mixer.h @@ -96,12 +96,15 @@ public: /** stop all currently playing sounds */ void stopAll(); - /** stop playing a specific sound */ - void stop(int index); + /** stop playing the given channel */ + void stop(int channel); - /** stop playing a specific sound */ + /** stop playing the sound with given ID */ void stopID(int id); + /** stop playing the channel for the given handle */ + void stopHandle(PlayingSoundHandle handle); + /** Start a new stream. */ int newStream(void *sound, uint32 size, uint rate, byte flags, uint32 buffer_size); @@ -114,7 +117,7 @@ public: /** Check whether any SFX channel is active.*/ bool hasActiveSFXChannel(); - /** Check whether the specified channel si active. */ + /** Check whether the specified channel is active. */ bool isActiveChannel(int index); /** bind to the OSystem object => mixer will be |